The invention relates to a method of controlling the gain of an amplifier, in particular for use in audio-signal-processing devices, a signal being applied to the amplifier and the output signal of the amplifier is digitized in an analog-to-digital converter. The invention also relates to an arrangement for carrying out the method.
Arrangements which process speech signals such as in particular speaker-identification systems and speech-recognition systems generally operate with a digitized speech signal. For this purpose the electrical analog speech signal must be applied from a microphone to an analog-to-digital converter, which converts the signal into a digital signal. Prior to its presentation to the converter, the speech signal from the microphone must be amplified to such an extent that the maximum range of said analog-to-digital converter is not, or hardly ever, exceeded and that the signal is not too small in comparison with the maximum range of the analog-to-digital converter. Since a speech-processing device is generally utilized by a multitude of persons who mostly speak with different loudnesses, an optimum drive of the analog-to-digital converter cannot be guaranteed if the gain of the amplifier, to which the speech signal from the microphone is applied, is fixed. For example, persons speaking in a loud voice may cause the maximum range of the converter to be exceeded, whereas persons speaking in a very soft voice produce a signal which may be partly drowned in the quantization of the converter.
In this respect it is to be noted that the scope of the invention is not limited to systems for speaker identification or speech recognition. Generally speaking, the invention may also be used in audio-signal-processing arrangements in general, such as, for example, audio cassette recorders and dictation equipment.
In order to adapt the gain factor of an amplifier to signals of different magnitude in such a way that a substantially unitary output signal is obtained, audio cassette recorders, for example, employ automatic gain control circuits of the analog type. However, these circuits have some drawbacks, namely their transient response in the case of loudness variations is generally very slow and, specifically, the gain during the recording of a speech signal is not constant. Especially if the conditions during recording of each signal must be well-defined and reproducible, the customary control circuits are unsatisfactory. However, for systems which process speech signals, such as automatic speaker identification systems and word-recognition systems, the conditions must be accurately reproducible. Automatic speaker identification systems generally employ a code word assigned to each individual speaker, for example a personal identification number, by means of which the speaker claims a specific identity. By means of the subsequently applied speech signal of this speaker it is verified whether the speaker actually has the identity claimed. Word-recognition systems are frequently also speaker-dependent i.e. before or when the speech-signal is applied it must be indicated which speaker is concerned. This need not necessarily be done by means of a number but it may also be effected by means of an acoustic signal. Moreover, in the case of speaker-identification and word-recognition systems used via the public telephone network, it is important to compensate for different degrees of attenuation.